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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-12-24 01:00:37 +00:00
rtspsrc: rework reconnect code
Use the same async code path to implement reconnects. Make sure we only post progress messages when doing async things.
This commit is contained in:
Wim Taymans
Mark Nauwelaerts
parent
c27c10f8f4
commit
2513207433
1 changed files with 96 additions and 67 deletions
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@ -3277,7 +3277,8 @@ gst_rtspsrc_push_event (GstRTSPSrc * src, GstEvent * event, gboolean source)
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}
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static GstRTSPResult
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gst_rtsp_conninfo_connect (GstRTSPSrc * src, GstRTSPConnInfo * info)
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gst_rtsp_conninfo_connect (GstRTSPSrc * src, GstRTSPConnInfo * info,
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gboolean async)
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{
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GstRTSPResult res;
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@ -3312,8 +3313,9 @@ gst_rtsp_conninfo_connect (GstRTSPSrc * src, GstRTSPConnInfo * info)
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if (!info->connected) {
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/* connect */
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GST_ELEMENT_PROGRESS (src, CONTINUE, "connect",
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("Connecting to %s", info->location));
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if (async)
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GST_ELEMENT_PROGRESS (src, CONTINUE, "connect",
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("Connecting to %s", info->location));
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GST_DEBUG_OBJECT (src, "connecting (%s)...", info->location);
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if ((res =
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gst_rtsp_connection_connect (info->connection,
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@ -3365,13 +3367,14 @@ gst_rtsp_conninfo_close (GstRTSPSrc * src, GstRTSPConnInfo * info,
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}
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static GstRTSPResult
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gst_rtsp_conninfo_reconnect (GstRTSPSrc * src, GstRTSPConnInfo * info)
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gst_rtsp_conninfo_reconnect (GstRTSPSrc * src, GstRTSPConnInfo * info,
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gboolean async)
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{
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GstRTSPResult res;
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GST_DEBUG_OBJECT (src, "reconnecting connection...");
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gst_rtsp_conninfo_close (src, info, FALSE);
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res = gst_rtsp_conninfo_connect (src, info);
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res = gst_rtsp_conninfo_connect (src, info, async);
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return res;
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}
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@ -3768,7 +3771,6 @@ invalid_length:
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static GstFlowReturn
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gst_rtspsrc_loop_udp (GstRTSPSrc * src)
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{
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gboolean restart = FALSE;
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GstRTSPResult res;
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GstRTSPMessage message = { 0 };
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gint retry = 0;
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@ -3809,7 +3811,6 @@ gst_rtspsrc_loop_udp (GstRTSPSrc * src)
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break;
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case GST_RTSP_EINTR:
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/* we got interrupted, see what we have to do */
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GST_DEBUG_OBJECT (src, "got interrupted: stop connection flush");
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goto interrupt;
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case GST_RTSP_ETIMEOUT:
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/* send keep-alive, ignore the result, a warning will be posted. */
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@ -3822,7 +3823,8 @@ gst_rtspsrc_loop_udp (GstRTSPSrc * src)
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* see what happens. */
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GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
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("The server closed the connection."));
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if ((res = gst_rtsp_conninfo_reconnect (src, &src->conninfo)) < 0)
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if ((res =
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gst_rtsp_conninfo_reconnect (src, &src->conninfo, FALSE)) < 0)
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goto connect_error;
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continue;
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@ -3868,17 +3870,79 @@ gst_rtspsrc_loop_udp (GstRTSPSrc * src)
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}
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}
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interrupt:
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/* we get here when the connection got interrupted */
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GST_OBJECT_LOCK (src);
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gst_rtspsrc_connection_flush (src, FALSE);
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GST_DEBUG_OBJECT (src, "we have command %d", src->loop_cmd);
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if (src->loop_cmd != CMD_RECONNECT)
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goto stopping;
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interrupt:
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{
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gst_rtsp_message_unset (&message);
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GST_DEBUG_OBJECT (src, "got interrupted: stop connection flush");
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gst_rtspsrc_connection_flush (src, FALSE);
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return GST_FLOW_WRONG_STATE;
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}
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connect_error:
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{
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gchar *str = gst_rtsp_strresult (res);
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GstFlowReturn ret;
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src->conninfo.connected = FALSE;
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if (res != GST_RTSP_EINTR) {
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GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ_WRITE, (NULL),
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("Could not connect to server. (%s)", str));
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g_free (str);
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ret = GST_FLOW_ERROR;
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} else {
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ret = GST_FLOW_WRONG_STATE;
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}
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return ret;
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}
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receive_error:
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{
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gchar *str = gst_rtsp_strresult (res);
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GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
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("Could not receive message. (%s)", str));
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g_free (str);
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return GST_FLOW_ERROR;
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}
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handle_request_failed:
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{
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gchar *str = gst_rtsp_strresult (res);
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GstFlowReturn ret;
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gst_rtsp_message_unset (&message);
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if (res != GST_RTSP_EINTR) {
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GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
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("Could not handle server message. (%s)", str));
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g_free (str);
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ret = GST_FLOW_ERROR;
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} else {
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ret = GST_FLOW_WRONG_STATE;
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}
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return ret;
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}
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server_eof:
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{
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GST_DEBUG_OBJECT (src, "we got an eof from the server");
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GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
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("The server closed the connection."));
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src->conninfo.connected = FALSE;
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gst_rtsp_message_unset (&message);
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return GST_FLOW_UNEXPECTED;
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}
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}
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static GstRTSPResult
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gst_rtspsrc_reconnect (GstRTSPSrc * src, gboolean async)
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{
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GstRTSPResult res = GST_RTSP_OK;
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gboolean restart;
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GST_DEBUG_OBJECT (src, "doing reconnect");
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GST_OBJECT_LOCK (src);
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/* only restart when the pads were not yet activated, else we were
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* streaming over UDP */
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restart = src->need_activate;
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src->flushing = FALSE;
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GST_OBJECT_UNLOCK (src);
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/* no need to restart, we're done */
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@ -3889,10 +3953,12 @@ interrupt:
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src->cur_protocols = GST_RTSP_LOWER_TRANS_TCP;
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/* pause to prepare for a restart */
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gst_rtspsrc_pause (src, FALSE, FALSE);
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if ((res = gst_rtspsrc_pause (src, FALSE, async)) < 0)
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goto done;
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/* close and cleanup our state */
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gst_rtspsrc_close (src, FALSE);
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if ((res = gst_rtspsrc_close (src, async)) < 0)
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goto done;
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/* see if we have TCP left to try. Also don't try TCP when we were configured
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* with an SDP. */
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@ -3907,52 +3973,17 @@ interrupt:
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gst_guint64_to_gdouble (src->udp_timeout / 1000000.0)));
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/* open new connection using tcp */
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if (!gst_rtspsrc_open (src, FALSE))
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if (gst_rtspsrc_open (src, async) < 0)
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goto open_failed;
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/* start playback */
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if (!gst_rtspsrc_play (src, &src->segment, FALSE))
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if (gst_rtspsrc_play (src, &src->segment, async) < 0)
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goto play_failed;
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done:
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return GST_FLOW_OK;
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return res;
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/* ERRORS */
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stopping:
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{
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GST_DEBUG_OBJECT (src, "we are stopping");
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GST_OBJECT_UNLOCK (src);
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return GST_FLOW_WRONG_STATE;
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}
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receive_error:
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{
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gchar *str = gst_rtsp_strresult (res);
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GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
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("Could not receive message. (%s)", str));
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g_free (str);
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return GST_FLOW_ERROR;
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}
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handle_request_failed:
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{
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gchar *str = gst_rtsp_strresult (res);
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GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
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("Could not handle server message. (%s)", str));
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g_free (str);
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gst_rtsp_message_unset (&message);
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return GST_FLOW_ERROR;
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}
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connect_error:
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{
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gchar *str = gst_rtsp_strresult (res);
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src->conninfo.connected = FALSE;
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GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ_WRITE, (NULL),
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("Could not connect to server. (%s)", str));
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g_free (str);
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return GST_FLOW_ERROR;
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}
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no_protocols:
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{
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src->cur_protocols = 0;
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@ -3973,15 +4004,6 @@ play_failed:
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GST_DEBUG_OBJECT (src, "play failed");
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return GST_FLOW_OK;
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}
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server_eof:
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{
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GST_DEBUG_OBJECT (src, "we got an eof from the server");
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GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
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("The server closed the connection."));
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src->conninfo.connected = FALSE;
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gst_rtsp_message_unset (&message);
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return GST_FLOW_UNEXPECTED;
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}
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}
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static void
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@ -4535,7 +4557,9 @@ receive_error:
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if (try == 0) {
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try++;
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/* if reconnect succeeds, try again */
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if ((res = gst_rtsp_conninfo_reconnect (src, &src->conninfo)) == 0)
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if ((res =
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gst_rtsp_conninfo_reconnect (src, &src->conninfo,
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FALSE)) == 0)
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goto again;
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}
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/* only try once after reconnect, then fallthrough and error out */
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@ -5068,7 +5092,7 @@ gst_rtspsrc_setup_streams (GstRTSPSrc * src, gboolean async)
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}
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if (src->conninfo.connection == NULL) {
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if (!gst_rtsp_conninfo_connect (src, &stream->conninfo)) {
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if (!gst_rtsp_conninfo_connect (src, &stream->conninfo, async)) {
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GST_DEBUG_OBJECT (src, "skipping stream %p, failed to connect", stream);
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continue;
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}
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@ -5469,7 +5493,7 @@ gst_rtspsrc_open_from_sdp (GstRTSPSrc * src, GstSDPMessage * sdp,
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src->conninfo.location = g_strdup (control);
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/* make a connection for this, if there was a connection already, nothing
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* happens. */
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if (gst_rtsp_conninfo_connect (src, &src->conninfo) < 0) {
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if (gst_rtsp_conninfo_connect (src, &src->conninfo, async) < 0) {
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GST_ERROR_OBJECT (src, "could not connect");
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}
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}
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@ -5527,7 +5551,7 @@ restart:
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goto no_url;
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src->tried_url_auth = FALSE;
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if ((res = gst_rtsp_conninfo_connect (src, &src->conninfo)) < 0)
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if ((res = gst_rtsp_conninfo_connect (src, &src->conninfo, async)) < 0)
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goto connect_failed;
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/* create OPTIONS */
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@ -6381,6 +6405,11 @@ gst_rtspsrc_thread (GstRTSPSrc * src)
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case CMD_LOOP:
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running = gst_rtspsrc_loop (src);
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break;
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case CMD_RECONNECT:
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ret = gst_rtspsrc_reconnect (src, FALSE);
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if (ret == GST_RTSP_OK)
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running = TRUE;
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break;
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default:
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break;
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}
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